About VOIP

Introduction to Voice-Over-IP

The Promise of Voice over IP
For over a decade now the prospect of using the internet to carry voice calls has been ‘next years technology’. Although there has not yet been any revolution in the way we route our phone calls, a number of enabling technologies, services and providers are now in place which can finally deliver a reliable, high-quality solution at very low cost.

Most businesses and individuals who are serious internet users now have un-timed and effectively un-limited connection to the internet. Users can spend all day downloading data from the other side of the world at no added cost. And yet, when those same users make a phone call they are charged by the minute, whether the call is local, national or international. In practice the data may well travel over exactly the same route, on the same wires, owned by the same people. Only the billing mechanism and price is different. Wouldn’t it be better for the end user if the telephone call went with the internet traffic with the attendant price saving?

Another attractive application for many businesses would be to connect home workers and sub-offices. The only on-going cost at each site would be the charge for an always-on internet connection. The remote sites could use the internet connection to log-in to the main office network and also run their telephones as extensions to the main office phone system.

A third use of Voice-over-IP technology is to replace the expensive telephone system that most companies require. The idea is to use existing computer hardware such as servers and Ethernet cabling to handle telephone traffic. Telephone system functions such as call-transfer and hold could be handled by software and telephone devices could just be plugged into a network point instead of dedicated wiring.

The three applications outlined above can be summarised as:

• Long-distance call routing.
• Point-to-point connections.
• In-house PBX systems.

The Hardware

VoIP Phones
VoIP Phones connect directly to your LAN via an RJ45 ethernet connection and provide quick and easy access to internet based telephony.
Gateways
A VoIP Gateway is a device which connects a telephone device or line to a computer network. On the computer connection side, devices may just have an ethernet connection or they may incorporate a cable-modem or ADSL modem. All the products available from IP:VOIP have 10/100 Mbps ethernet ports for their network side connection.
At the telephone side, some of the IP:VOIP products provide standard analogue (also called PSTN or POTS) connections. These connections come in two flavours:

• FXS use with devices like phones, fax machines or PBX trunk ports
  
• FX0 connect to a trunk line from BT or a PBX extension.

Other IP:VOIP products provide ISDN2e (BRI) and ISDN30e (PRI / E1) connectivity on the telephone side.

The Channel

In principle you can use a VoIP gateway to communicate with anyone else on the internet who is similarly equipped, or has software to drive their PC/Soundcard. For best performance it is preferred that both ends have some form of broadband connectivity as a minimum.

A more common use is to connect two or more sites for free calls between the sites. There may already be a data-link between the sites or the prospect of free-calls may be the spur to set this up.

Each telephone conversation requires a channel of less than 10k, so any data-link from 64k up would be reasonable as long as it is fairly stable, has a small delay and is not already congested.

Suitable choices for the channel are

• Direct wired/RF/IR/Microwave ethernet connection
  
• Leased Line / IP VPN
  
• ISDN/ADSL/CableModem dialup/FRIACO accounts with suitable ISP

Of these the most 'asked about' is the last. Here is a list of requirements for such a dial-up account system

Must have fixed (though not necessarily public) IP
ISPs must not block VoIP protocols (some ISP's may have a vested interest in not allowing you to do telephone calls via the internet)
We can help with arranging suitable fixed monthly 'always-on' accounts if required

The protocols and call-routing
The first standard for videoconferencing, telephony and other multimedia use of computer networks was H.323, created under the auspices of the ITU in 1996 and augmented many times since. The last major revision was version 4 in 2000. The standard has also acquired H.248 to control gateway functions and H.235 for its security framework; future additions may include H.460, for adding extensions such as number portability across locations, and the H.5xx series for mobile users. Although much in the protocol family leans towards a world of 64k pipes through ATM networks, it was originally designed to run over LANs and has grown out to encompass WANs subsequently. Calls go between endpoints -- phones, computers, video-conferencing systems -- while the media and signalling are handled by gateways. Optionally, a gatekeeper will do call management, routing and address resolution.

SIP -- the Session Initiation Protocol -- is relatively new, only coming to prominence over the past couple of years. Strictly speaking, it isn't a telephony protocol; instead, it does the call set-up, error handling and inter-process signalling that goes along with any point-to-point connection. When used for telephony it most often uses the same underlying streaming protocol, RTP, as H.323. SIP is -- or was, when it was born -- as simple as H.323 is complex, a text-based protocol that finds the recipient of a call, checks that it has capabilities congruent with the caller's, and then lets other protocols take care of the details of data transfer, security and so on. It was created by the IETF, and like H.323 has grown a number of related protocols -- SIP-CPL, the Call Processing Language, is a scripting language based on XML tags, SIP-CGI defines how server-resident scripts can communicate with applications, among others. Unlike H.323, SIP moves a lot of the work of call management and routing out among different parts of the network.

H.323 vs SIP - click here for comparison table